In recent years, the Voice over IP (VoIP) technology was developed in which a phone call as an example of a use signal is sent via an IP-based network (IP network, internet protocol network) such as the Internet, for example. By sending the signal via such a network instead of a conventional long distance carrier, it is possible to reduce the costs involved for such a call.
A basic structure of a communication system using the above technology is shown in FIG. 1. For the purpose of the following description, the devices shown on the left side of the IP network (transmitting network) 4 in FIG. 1 is referred to as the near-end side, while the right side is referred to as the far-end side.
A mobile phone 1 as a first communication device is connected to a first network control device 2 for controlling a first network (near-end network) to which the mobile phone 1 is connected. The first network control device 2 is, for example, a mobile services switching center (MSC). A speech signal is sent, for example at a bit rate of, e.g., 64 kbps from the first network control device 1 to a first gateway 3 which connects the near-end network with the IP network 4. In order to achieve capacity saving on the IP link, the speech is compressed in the gateway.
This compression is performed by a codec (coder-decoder, transcoder, code converter) arranged in the first gateway 3. A typical compression ratio for speech is, for example, 8:1. Since the function of the codec itself is not important to the present invention, a detailed description thereof is omitted here.
The speech signal is compressed, for example, to a bit rate of 8 kbps. The compressed speech signal is sent via the IP network 4 to a second gateway 5. This second gateway also comprises a codec (coder-decoder). However, this codec 5a decompresses the compressed signal received from the IP network 4 to restore the original rate (i.e., in the above example, 64 kbps). The decompressed speech signal US is sent to a second network control device 6 for controlling a second network (far-end network) to which a phone 7 as a second communication device is connected. The second network control device 6 can be a mobile services switching center (MSC) in case the phone 7 is a mobile phone or a fixed services switching center (FSC) in case the phone 7 is a fixed phone. The second network control device 6 sends the signal to the destination phone 7.
As described above, the speech signal is compressed and decompressed. If also tone signals such as DTMF-tones (dual tone multi frequency) would be compressed and decompressed in the same way as the speech, these tones would be corrupted. The tones may sound sufficient to human ears but they are probably degraded out of specifications of tone-managed services.
Hence, usually a DTMF-tone detection is implemented in the gateways to bypass the codec.
Such a DTMF-tone detection is described hereinafter with reference to FIG. 2. In FIG. 2, the two gateways 3 and 5 of FIG. 1 are shown in more detail. For simplifying the description, the network, over which the signals are transmitted, is not shown.
The first gateway 3 comprises a codec 3a, a tone detection means 3b and a tone generation means 3c. Likewise, the second gateway 5 comprises a codec 5a, a tone detection means 5b and a tone generation means 5c. 
Reference character TS denotes a DTMF tone signal. The DTMF tone signal TS is detected by the tone detection means 3b of the near-end gateway 3. For sending the tone to the far-end gateway 5, a signaling message CS (control signal) is used. This signaling message can have an appropriate format as defined in standards. In the far-end gateway 5, the signaling message CS is received by the tone generation means 5c, by means of which the DTMF tone signal TS is generated again and supplied to the speech signal line. During the tone transfer, the speech channel may be idle.
In the following, a description is given as to how DTMF-tones are generated by using mobile phones with respect to FIGS. 2 and 3.
In FIG. 3, reference character 1 denotes a mobile-phone. When a key of this mobile phone 1 is pressed, the signaling message CS is sent to the first network control device 2, in which the corresponding DTMF tone signal TS is generated by a tone generating means 2a. The tone signal TS is included into the speech signal which is denoted by reference character US (use signal). That is, by including the DTMF tone signal TS in the speech signal US, the speech signal US is affected by the DTMF tone signal TS.
The speech signal US is sent to the first gateway 3, in which the DTMF tone signal is detected and converted by the tone generation means 3b to a signaling message again, as shown in FIG. 2. Furthermore, the speech signal US is compressed in a codec 3a of the first gateway 3 shown in FIG. 2. The compressed speech signal is denoted with reference character USC (compressed use signal).
The compressed speech signal USC and the signaling message CS are sent via the IP network to the second gateway 5. There, the signaling message CS is used to generate the tone into the speech channel, as described with respect to FIG. 2. This signal is sent to the second network control device 6.
Hence, in this conventional communication system, the signaling message is converted into a tone signal and then again converted into a signaling message which causes the tone generating means 5c in the second gateway 5 to generate a corresponding tone. This results in the drawbacks of a large hardware requirement and of disturbances of the use signal which may include a TFO (tandem free operation) stream in case of a mobile-to-mobile call.
In the following, the TFO (tandem free operation) is described in more detail. Usually in a mobile-to-mobile call the speech is encoded in the first mobile phone 1 to 8 kbps and then decoded in a codec (transcoder, code converter) before the network control device (for example, MSC) back to 64 kbps. In the first network control device 2, the 64 kbps call is turned back and it is encoded again in a codec to 8 kbps. This signal is sent to the receiving side and decoded in the same manner in the other mobile phone back to 64 kbps.
The tandem free operation (TFO) makes it possible to omit the unnecessary decoding/encoding phases in codecs. For compatibility of network control devices being capable of performing the TFO and network control devices not being able to perform the TFO, both formats (8 and 64 kbps) are sent from the transcoder towards the MSC, wherein the encoded parameters are inserted in the least significant bits of a 64 kbps stream. This insertion degrades the quality of the 64 kbps stream only to a small extent.
However, in such a case, the tone generation in the first network control device 2 as described above corrupts the TFO stream inside the 64 kbps stream, such that the TFO stream synchronisation can get lost.
Document WO-A-97 50262 dislcoses a communication system and method as defined in the preambles of claims 1 and 23. A use signal and a control signal are transmitted separately via a part of the communication path.
Document EP-A-0 589 619 discloses an architecture for a wireless telecommunication system in which voice channels and data links are provided separately between a multiplexer connected to a private branch exchange (PBX) and a cell site.